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• ffmpeg removes dead space in video but not audio

  10 juillet 2015, par Peter Becich

  I’m transferring a VHS tape on Ubuntu with ffmpeg, somagic-capture, the line-in jack and an EasyCap dongle.

  There is a big issue with audio drift that, I believe, is caused by ffmpeg removing dead space from the video stream but not the audio stream.

  The capture utility creates a stream :

  somagic-capture --ntsc -c --luminance=2 --lum-aperture=3 \
  
  2> $SOMAGIC_LOG |

  Which is piped into ffmpeg. ffmpeg also captures from alsa :

   ffmpeg -pixel_format uyvy422 -s:v 720x480 \
  
  -framerate 29.97 -f rawvideo -i -  -f alsa -thread_queue_size 1024 \
  
  -i hw:0,0 -vf scale=w=640:h=480 -vcodec libx264 -preset ultrafast \
  
  -shortest -c:a libfdk_aac -b:a 256k $OUTFILE

  Is it true that ffmpeg by default removes dead space from a stream ?

  If so, would ffmpeg trim both streams even if only one is "dead" — suppose the video is momentarily blank while the audio is noisy.

  If not, are these two streams incompatible with each other ? Is the ALSA input (line-in jack) on a different clock than the EasyCap video dongle ?

  Input #0, rawvideo, from 'pipe:':
  
    Duration: N/A, start: 0.000000, bitrate: 165722 kb/s
  
      Stream #0:0: Video: rawvideo (UYVY / 0x59565955), uyvy422, 720x480, 
  
      165722 kb/s, 29.97 tbr, 29.97 tbn, 29.97 tbc
  
  Guessed Channel Layout for  Input Stream #1.0 : stereo
  
  
  
  Input #1, alsa, from 'hw:0,0':
  
    Duration: N/A, start: 1436492353.282796, bitrate: 1536 kb/s
  
    Stream #1:0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s

  I think the -shortest flag only concerns killing the ffmpeg process.

  My prior, more general question didn’t solve the problem : http://video.stackexchange.com/questions/14809/sync-up-ffmpeg-rawvideo-recording-and-audacity-alsa-recording

  Thanks for any advice !

  The full script :

  #!/bin/sh
  
  
  
  PIPE=/tmp/somagic-pipe
  
  OUTFILEDIR=~/easycap/Videos/
  
  LOGDIR=~/.somagic-log/
  
  NOW=`date +"%m_%d_%Y_%H_%M_%S"`
  
  
  
  OUTFILE=${OUTFILEDIR}fpv_${NOW}_video.mp4
  
  
  
  mkdir $LOGDIR
  
  
  
  FFMPEG_LOG=${LOGDIR}ffmpeg_video.log
  
  SOMAGIC_LOG=${LOGDIR}somagic_video.log
  
  MPLAYER_LOG=${LOGDIR}mplayer_video.log
  
  
  
  rm $PIPE >/dev/null 2>&1
  
  rm $OUTFILE >/dev/null 2>&1
  
  
  
  rm $FFMPEG_LOG
  
  rm $SOMAGIC_LOG
  
  rm $MPLAYER_LOG
  
  
  
  mkfifo $PIPE >/dev/null 2>&1
  
  
  
  somagic-capture --ntsc -c --luminance=2 --lum-aperture=3 \
  
  2> $SOMAGIC_LOG | ffmpeg -pixel_format uyvy422 -s:v 720x480 \
  
  -framerate 29.97 -f rawvideo -i -  -f alsa -thread_queue_size 1024 \
  
  -i hw:0,0 -vf scale=w=640:h=480 -vcodec libx264 -preset ultrafast \
  
  -shortest -c:a libfdk_aac -b:a 256k $OUTFILE
  
  
  
  rm $PIPE >/dev/null 2>&1 

  Modified from here : https://gist.github.com/Brick85/0b327ac2d3d45e23ed33

• Low latency video streaming on android

  17 mai 2021, par Louis Blenner

  I'd like to be able to stream the video from my webcam to an Android app with a latency below 500ms, on my local network.

  


  To capture and send the video over the network, I use ffmpeg.

  


  ffmpeg -f v4l2 -i /dev/video0 -preset ultrafast -tune zerolatency -vcodec libx264 -an -vf format=yuv420p -f mpegts  udp://192.168.1.155:5000


  


  This command takes the webcam as an input, convert it and send it to a device using the mpegts protocol.

  


  I am able to read the video on another PC with a latency below 500 ms, using commands like

  


  gst-launch-1.0 -v udpsrc port=5000 ! video/mpegts ! tsdemux ! h264parse ! avdec_h264 ! fpsdisplaysink sync=false


  


  or

  


  mpv udp://0.0.0.0:5000 --no-cache --untimed --no-demuxer-thread --video-sync=audio --vd-lavc-threads=1 


  


  So it is possible to have this range of latency.
  
I'd like to have the same thing on Android.

  


  Here are my tries to do that.

  


  Exoplayer

  


  After looking at the different players available on Android studio, it seems like Exoplayer is the go-to choice.
  
I tried different options indicated in the live-streaming documentation, but I always end up with a stream taking seconds to start and with a latency of seconds.
  
I tried to add a Button to seek to the default position of the windows, but it results in a loading of several seconds.

  


  DefaultExtractorsFactory extractorsFactory =
                new DefaultExtractorsFactory()
                        .setTsExtractorFlags(DefaultTsPayloadReaderFactory.FLAG_IGNORE_AAC_STREAM);

        player = new SimpleExoPlayer.Builder(this)
                .setMediaSourceFactory(
                        new DefaultMediaSourceFactory(this, extractorsFactory))
                .setLoadControl(new DefaultLoadControl.Builder()
                        .setBufferDurationsMs(DefaultLoadControl.DEFAULT_MIN_BUFFER_MS, DefaultLoadControl.DEFAULT_MAX_BUFFER_MS, 200, 200)
                        .build())
                .build();
        MyPlayerView playerView = findViewById(R.id.player_view);
        // Bind the player to the view.
        playerView.setPlayer(player);
        // Build the media item.
        MediaItem mediaItem = new MediaItem.Builder()
                .setUri(Uri.parse("udp://0.0.0.0:5000"))
                .setLiveMaxOffsetMs(500)
                .setLiveTargetOffsetMs(0)
                .setLiveMinOffsetMs(0)
                .build();
        // Set the media item to be played.
        player.setMediaItem(mediaItem);
        // Prepare the player.
        player.setPlayWhenReady(true);
        player.prepare();
        //player.seekToDefaultPosition();


  


  This issue is about the same issue and the conclusion was that Exoplayer was not fit for this use case.

  


  


  I'll be honest, ultra low-latency like this isn't ExoPlayer's main use-case

  


  


  Vlc

  


  Another try was to use the Vlc library.
  
But I was unable to have the same low latency stream as with the two previous example with Vlc.
  
I tried changing the preferences of Vlc to stream as fast as possible.

  


  Input/Codecs -> x264 preset: ultrafast - zerolatency
Input/Codecs -> Access Module: UDP input
Input/Codecs -> Clock Jitter: 500
Audio: disable audio


  


  I also tried reducing the different buffers.
  
However, I still have a latency of more than 1 seconds with that.

  


  Gstreamer

  


  Another try was to create a react-native project to use the different players available here.
  
One player that seemed promising was react-native-gstreamer because it uses gstreamer which is able to stream with low latency (gst-launch command).
  
But the library is now outdated.

  


  Question

  


  There were other tries, but none were successful.
  
Is there a problem with one of my approaches ?
  
And if not, Is there a player on Android (that I missed) which is able to achieve low latency stream like gstream or mpv on linux ?

  


• ffplay 461 Unsupported transport

  28 octobre 2017, par Cerato

  Please help me to understand what I am doing wrong.

  I am executing the following command for making a live stream

  Executing: powershell gst-launch-1.0 filesrc location=D:/testVODs/sample.mkv ! matroskademux name=dmx dmx. ! h264parse config-interval=1 ! queue ! rtspclientsink profiles=avpf protocols=udp location="rtsp://192.168.1.1:8554/8c83bea4-b49b-4776-9c0f-d11e9e8898f0/1" name=rtsp_out dmx. ! opusparse ! queue ! rtsp_out.

  and in the result, I see that the execution ended after 00:01:21 which means that the stream is ended successfully after my video ended :

  Setting pipeline to PAUSED ...
  
  Pipeline is PREROLLING ...
  
  Progress: (open) Opening Stream
  
  Progress: (connect) Connecting to rtsp://192.168.1.1:8554/8c83bea4-b49b-4776-9c0f-d11e9e8898f0/1
  
  Progress: (open) Retrieving server options
  
  Progress: (open) Opened Stream
  
  Pipeline is PREROLLED ...
  
  Setting pipeline to PLAYING ...
  
  New clock: GstSystemClock
  
  Progress: (request) Sending RECORD request
  
  Progress: (record) Sending server stream info
  
  Progress: (request) Sending RECORD request
  
  Progress: (request) SETUP stream 1
  
  Progress: (request) SETUP stream 0
  
  Progress: (record) Starting recording
  
  Got EOS from element "pipeline0".
  
  Execution ended after 0:01:21.318315052
  
  Setting pipeline to PAUSED ...
  
  Setting pipeline to READY ...
  
  Setting pipeline to NULL ...
  
  Freeing pipeline ...

  at the same time I am trying to play the stream using the following request :

  Executing: powershell ffplay -rtsp_transport http "rtsp://192.168.1.1:8554/live/8c83bea4-b49b-4776-9c0f-d11e9e8898f0/1"

  but I have the following response :

  ffplay started on 2017-10-27 at 18:52:13
  
  Report written to "D:/testVODs/Stream1to.log"
  
  ffplay version 3.4 Copyright (c) 2003-2017 the FFmpeg developers
  
    built with gcc 7.2.0 (GCC)
  
    configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-bzlib --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-cuda --enable-cuvid --enable-d3d11va --enable-nvenc --enable-dxva2 --enable-avisynth --enable-libmfx
  
    libavutil      55. 78.100 / 55. 78.100
  
    libavcodec     57.107.100 / 57.107.100
  
    libavformat    57. 83.100 / 57. 83.100
  
    libavdevice    57. 10.100 / 57. 10.100
  
    libavfilter     6.107.100 /  6.107.100
  
    libswscale      4.  8.100 /  4.  8.100
  
    libswresample   2.  9.100 /  2.  9.100
  
    libpostproc    54.  7.100 / 54.  7.100
  
      nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0   
  
      nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0   
  
      nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0   
  
      nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0   
  
      nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0   
  
      nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0   
  
      nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0   
  
      nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0   
  
      nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0   
  
      nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0   
  
      nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0   
  
      nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0   
  
      nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0   
  
  [rtsp @ 000001930a05e500] method SETUP failed: 461 Unsupported transport
  
  rtsp://192.168.1.1:8554/live/8c83bea4-b49b-4776-9c0f-d11e9e8898f0/1: Unknown error

  why method SETUP failed as 461 Unsupported transport and how it can be solved ?

  I tried to use different ffplay, ffmpeg commands, but all of them lead to the same result.
  If I do not use rtsp_trasport command I have the error [udp @ 000001d1ab52c300] ’circular_buffer_size’ option was set but it is not supported on this build (pthread support is required)
  and the same 461 Unsupported transport

  Before I added "profiles=avpf" to my stream creation request I managed to play it successfully with "rtsp_transport tcp".
  But now profiles=avpf is required and I need to understand how to deal with it